Voice Quality Issues and Solutions
  • 29 Apr 2021
  • 5 Minutes to read
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Voice Quality Issues and Solutions

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Article Summary

Quality and clarity of voice are very important for users. There is useful information in this document about some of the problems you may encounter in voice communication. There are two types of voice problems, problems on the user side and problems on the system side. In the first part, some corrections to be made by the user are explained. In the second part, the use of Tegsoft Speed Test application, where you can get information about your connection speed, is shown. Speed Test is a feature of Tegsoft application designed for users. There are couple of reasons that may cause voice quality issues.

In addition, the analysis of problems to be experienced in the system and possible solutions are explained.

Conditions to be Controlled by The User

Users should do the first check on their own computers and network connections. If the problem continues after these checks, they should do the technical checks in the content of the article.

  • Headset quality is important, we usually recommend good brand headsets with USB type for best quality.
  • User local environment, network quality and internet speed are important. (like PC performance, network quality)
  • Network quality between Tegsoft and user's computer. If there is VPN related or any other network issues, that needs to be monitored and fixed for best quality.
  • The network quality between the voice provider and Tegsoft should be checked. If there is a problem, it should be fixed.
  • Voice provider service quality is very important.
  • It is checked that the necessary permissions are given for the headset in the webservice settings.

Tegsoft solution has some internal tools to monitor and identify voice quality issues. POC is the only way to activate those tools and check results.

Your Connection Speed with Tegsoft Server

If your internet is slow, your internet connection is bad, your entries on the system are not recorded, you cannot hear the customer or the customer cannot hear you, you can use Speedtest which is the new application of Tegsoft to check your connection.
Go to your URL bar and delete the part after the TegsoftCloud.com and write Speedtest as shown on the screen, and press the enter key. After accessing this page, you can start the test.
After starting the test, wait until the results appear. After starting the test, wait until the results appear. The results should take Upload and Dowland values and the jitter value should be small. These values may vary depending on your internet connection speed.

Download: Download speed for data downloaded from the internet.
Upload: Upload speed of data from your computer to another computer over the internet.
Jitter: The time it takes for the signal to leave the server and arrive at your computer. A large jitter value means delay in the incoming signal.

Screen Shot 2021-04-28 at 13.49.18.png

How to Analyze Voice Quality Issues?

Voice is transmitted from peer to peer via the network layer. There may be the following issues listed here during voice communication.



A B < ------->


A Gateway / Firewall VOIP <-------> <------------>

One-way or No Voice

Reason for this result is usually because of non-transmitted RTP packets.

This case is usually observed in the External voice transmission case. It is not common to be faced a no voice issue if the peers are in the same network.

For example,
If two users have a voice problem while talking between themselves, the problem is internal.

Possible Cause
Hardware related issues may cause this result. Please check the microphone, headset, phone device.
There may be a network issue between switches that carries traffic between A and B.

This issue usually called NAT problem. As you can notice from the example SIP package below, peers send audio I/O port numbers and IP addresses to the destination. If A sends the local IP address to the destination VOIP. VOIP will not be able to send RTP packages to A.

For example,
If users have a voice problem in an inbound or outbound call, they do not hear the voice of the customers, or their voice is not going to the customer, the problem is external.

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP;branch=z9hG4bK3fad5b24
From: "Eray GURSOY" <sip:[email protected]>;tag=682c7b5d8f2c884f70b69c59-68ee017e To: <sip:[email protected]>
Call-ID: [email protected]
Max-Forwards: 70
Session-ID: 2c5e350000105000a000682c7b5d8f2c;remote=00000000000000000000000000000000
Date: Fri, 15 Nov 2019 08:15:52 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP8845/11.0.1
Contact: <sip:[email protected]:5060;user=phone;transport=udp>;+u.sip!devicename.ccm.cisco.com ="SEP682C7B5D8F2C" Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO Remote-Party-ID: "Eray GURSOY" <sip:[email protected]>;party=calling;id- type=subscriber;privacy=off;screen=yes Supported: replaces,join,sdp-anat,norefersub,resource- priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco- escapecodes,X-cisco-service- control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1 Allow-Events: kpml,dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Authorization: Digest username="1901",realm="asterisk",uri="sip:[email protected];user=phone",response="a6 d395930c2976c0b16a633fef2129b0" ,nonce="73456cb7",algorithm=MD5
Content-Length: 352
Content-Type: application/sdp
Content-Disposition: session;handling=optional
o=Cisco-SIPUA 21523 0 IN IP4 s=SIP Call
t=0 0
m=audio 27480 RTP/AVP 0 8 116 18 101 c=IN IP4
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-15 

Possible Cause
Network definitions of Tegsoft server may be incorrect, please check NAT definitions from PBX Management / Server Settings / NAT section.
Firewall or the gateway may perform NAT operation either none or partial.

Missing Words or Some Parts in Speech

This case is noticed when you hear a speech like below.
Original speech → “I am telling you some details here about the last meeting.”
Experienced speech → “I ... tell.. you ...e details ... about the last meeting.”
RTP transmission is designed to overcome data loss. To overcome data loss peers generate statistics and use jitter buffers to deliver proper speech transmission. When jitter buffer is not enough or when the transmission is that bad peers start to lose some parts of the speech.

Possible Cause
Problem definition is “Network transmission issue”. Please check network quality and bandwidth or codec.

Digitalization in Speech

This case is noticed when you hear robotic voices or speech.
When analog (voice) to IP (RTP Package) conversion is failing because of resources on the PBX server (CPU, DSP) you may face this issue. Tegsoft support 90 channel with SMB hardware, 400 channel for Intel CPU server, 10,000 channel for IBM Power server.

Hearing Buzzy or Deep Voice During Communication

Invalid coded during transmission will cause this issue. Please check codec capability between peers.

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